forked from mikebrady/shairport-sync
-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathaudio_alsa.c
1040 lines (927 loc) · 37 KB
/
audio_alsa.c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* libalsa output driver. This file is part of Shairport.
* Copyright (c) Muffinman, Skaman 2013
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or
* sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
#define ALSA_PCM_NEW_HW_PARAMS_API
#include "audio.h"
#include "common.h"
#include <alsa/asoundlib.h>
#include <math.h>
#include <memory.h>
#include <pthread.h>
#include <stdio.h>
#include <unistd.h>
static void help(void);
static int init(int argc, char **argv);
static void deinit(void);
static void start(int i_sample_rate, int i_sample_format);
static void play(short buf[], int samples);
static void stop(void);
static void flush(void);
int delay(long *the_delay);
static void volume(double vol);
void do_volume(double vol);
void do_mute(int request);
static void linear_volume(double vol);
static void parameters(audio_parameters *info);
static void mute(int do_mute);
static double set_volume;
static int output_method_signalled = 0;
audio_output audio_alsa = {.name = "alsa",
.help = &help,
.init = &init,
.deinit = &deinit,
.start = &start,
.stop = &stop,
.flush = &flush,
.delay = &delay,
.play = &play,
.mute = NULL, // a function will be provided if it can, and is allowed to, do hardware mute
.volume = NULL, // a function will be provided if it can do hardware volume
.parameters = ¶meters};
static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
static unsigned int desired_sample_rate;
static enum sps_format_t sample_format;
static snd_pcm_t *alsa_handle = NULL;
static snd_pcm_hw_params_t *alsa_params = NULL;
static snd_ctl_t *ctl = NULL;
static snd_ctl_elem_id_t *elem_id = NULL;
static snd_mixer_t *alsa_mix_handle = NULL;
static snd_mixer_elem_t *alsa_mix_elem = NULL;
static snd_mixer_selem_id_t *alsa_mix_sid = NULL;
static long alsa_mix_minv, alsa_mix_maxv;
static long alsa_mix_mindb, alsa_mix_maxdb;
static char *alsa_out_dev = "default";
static char *alsa_mix_dev = NULL;
static char *alsa_mix_ctrl = "Master";
static int alsa_mix_index = 0;
static int hardware_mixer = 0;
static int has_softvol = 0;
static int volume_set_request = 0; // set when an external request is made to set the volume.
int mute_request_pending = 0; // set when an external request is made to mute or unmute.
int overriding_mute_state_requested = 0; // 1 = mute; 0 = unmute requested
static snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *,
snd_pcm_uframes_t) = snd_pcm_writei;
static int play_number;
static int64_t accumulated_delay, accumulated_da_delay;
int alsa_characteristics_already_listed = 0;
static snd_pcm_uframes_t period_size_requested, buffer_size_requested;
static int set_period_size_request, set_buffer_size_request;
static void help(void) {
printf(" -d output-device set the output device [default*|...]\n"
" -m mixer-device set the mixer device ['output-device'*|...]\n"
" -c mixer-control set the mixer control [Master*|...]\n"
" -i mixer-index set the mixer index [0*|...]\n"
" *) default option\n");
}
int open_mixer() {
if (hardware_mixer) {
debug(2, "Open Mixer");
int ret = 0;
snd_mixer_selem_id_alloca(&alsa_mix_sid);
snd_mixer_selem_id_set_index(alsa_mix_sid, alsa_mix_index);
snd_mixer_selem_id_set_name(alsa_mix_sid, alsa_mix_ctrl);
if ((snd_mixer_open(&alsa_mix_handle, 0)) < 0)
die("Failed to open mixer");
debug(3, "Mixer device name is \"%s\".", alsa_mix_dev);
if ((snd_mixer_attach(alsa_mix_handle, alsa_mix_dev)) < 0)
die("Failed to attach mixer");
if ((snd_mixer_selem_register(alsa_mix_handle, NULL, NULL)) < 0)
die("Failed to register mixer element");
ret = snd_mixer_load(alsa_mix_handle);
if (ret < 0)
die("Failed to load mixer element");
debug(3, "Mixer Control name is \"%s\".", alsa_mix_ctrl);
alsa_mix_elem = snd_mixer_find_selem(alsa_mix_handle, alsa_mix_sid);
if (!alsa_mix_elem)
die("Failed to find mixer element");
return 1;
} else {
return 0;
}
}
void close_mixer() {
if (alsa_mix_handle) {
snd_mixer_close(alsa_mix_handle);
alsa_mix_handle=NULL;
}
}
static int init(int argc, char **argv) {
pthread_mutex_lock(&alsa_mutex);
// debug(2,"audio_alsa init called.");
const char *str;
int value;
double dvalue;
set_period_size_request = 0;
set_buffer_size_request = 0;
config.alsa_use_playback_switch_for_mute = 1;
config.audio_backend_latency_offset = 0;
config.audio_backend_buffer_desired_length = 0.15;
// get settings from settings file first, allow them to be overridden by
// command line options
if (config.cfg != NULL) {
/* Get the desired buffer size setting. */
if (config_lookup_int(config.cfg, "alsa.audio_backend_buffer_desired_length", &value)) {
if ((value < 0) || (value > 66150)) {
inform("The setting alsa.audio_backend_buffer_desired_length is deprecated. "
"Use alsa.audio_backend_buffer_desired_length_in_seconds instead.");
die("Invalid alsa audio backend buffer desired length \"%d\". It "
"should be between 0 and "
"66150, default is 6615",
value);
} else {
inform("The setting alsa.audio_backend_buffer_desired_length is deprecated. "
"Use alsa.audio_backend_buffer_desired_length_in_seconds instead.");
config.audio_backend_buffer_desired_length = 1.0 * value / 44100;
}
}
/* Get the desired buffer size setting. */
if (config_lookup_float(config.cfg, "alsa.audio_backend_buffer_desired_length_in_seconds",
&dvalue)) {
if ((dvalue < 0) || (dvalue > 1.5)) {
die("Invalid alsa audio backend buffer desired time \"%f\". It "
"should be between 0 and "
"1.5, default is 0.15 seconds",
dvalue);
} else {
config.audio_backend_buffer_desired_length = dvalue;
}
}
/* Get the latency offset. */
if (config_lookup_int(config.cfg, "alsa.audio_backend_latency_offset", &value)) {
if ((value < -66150) || (value > 66150)) {
inform("The setting alsa.audio_backend_latency_offset is deprecated. "
"Use alsa.audio_backend_latency_offset_in_seconds instead.");
die("Invalid alsa audio backend buffer latency offset \"%d\". It "
"should be between -66150 and +66150, default is 0",
value);
} else {
inform("The setting alsa.audio_backend_latency_offset is deprecated. "
"Use alsa.audio_backend_latency_offset_in_seconds instead.");
config.audio_backend_latency_offset = 1.0 * value / 44100;
}
}
/* Get the latency offset. */
if (config_lookup_float(config.cfg, "alsa.audio_backend_latency_offset_in_seconds", &dvalue)) {
if ((dvalue < -1.0) || (dvalue > 1.5)) {
die("Invalid alsa audio backend buffer latency offset time \"%f\". It "
"should be between -1.0 and +1.5, default is 0 seconds",
dvalue);
} else {
config.audio_backend_latency_offset = dvalue;
}
}
/* Get the Output Device Name. */
if (config_lookup_string(config.cfg, "alsa.output_device", &str)) {
alsa_out_dev = (char *)str;
}
/* Get the Mixer Type setting. */
if (config_lookup_string(config.cfg, "alsa.mixer_type", &str)) {
inform("The alsa mixer_type setting is deprecated and has been ignored. "
"FYI, using the \"mixer_control_name\" setting automatically "
"chooses a hardware mixer.");
}
/* Get the Mixer Device Name. */
if (config_lookup_string(config.cfg, "alsa.mixer_device", &str)) {
alsa_mix_dev = (char *)str;
}
/* Get the Mixer Control Name. */
if (config_lookup_string(config.cfg, "alsa.mixer_control_name", &str)) {
alsa_mix_ctrl = (char *)str;
hardware_mixer = 1;
}
/* Get the disable_synchronization setting. */
if (config_lookup_string(config.cfg, "alsa.disable_synchronization", &str)) {
if (strcasecmp(str, "no") == 0)
config.no_sync = 0;
else if (strcasecmp(str, "yes") == 0)
config.no_sync = 1;
else
die("Invalid disable_synchronization option choice \"%s\". It should be \"yes\" or \"no\"");
}
/* Get the mute_using_playback_switch setting. */
if (config_lookup_string(config.cfg, "alsa.mute_using_playback_switch", &str)) {
if (strcasecmp(str, "no") == 0)
config.alsa_use_playback_switch_for_mute = 0;
else if (strcasecmp(str, "yes") == 0)
config.alsa_use_playback_switch_for_mute = 1;
else
die("Invalid mute_use_playback_switch option choice \"%s\". It should be \"yes\" or \"no\"");
}
/* Get the output format, using the same names as aplay does*/
if (config_lookup_string(config.cfg, "alsa.output_format", &str)) {
if (strcasecmp(str, "S16") == 0)
config.output_format = SPS_FORMAT_S16;
else if (strcasecmp(str, "S24") == 0)
config.output_format = SPS_FORMAT_S24;
else if (strcasecmp(str, "S24_3LE") == 0)
config.output_format = SPS_FORMAT_S24_3LE;
else if (strcasecmp(str, "S24_3BE") == 0)
config.output_format = SPS_FORMAT_S24_3BE;
else if (strcasecmp(str, "S32") == 0)
config.output_format = SPS_FORMAT_S32;
else if (strcasecmp(str, "U8") == 0)
config.output_format = SPS_FORMAT_U8;
else if (strcasecmp(str, "S8") == 0)
config.output_format = SPS_FORMAT_S8;
else
die("Invalid output format \"%s\". It should be \"U8\", \"S8\", \"S16\", \"S24\", "
"\"S24_3LE\", \"S24_3BE\" or "
"\"S32\"",
str);
}
/* Get the output rate, which must be a multiple of 44,100*/
if (config_lookup_int(config.cfg, "alsa.output_rate", &value)) {
debug(1, "Value read for output rate is %d.", value);
switch (value) {
case 44100:
case 88200:
case 176400:
case 352800:
config.output_rate = value;
break;
default:
die("Invalid output rate \"%d\". It should be a multiple of 44,100 up to 352,800", value);
}
}
/* Get the use_mmap_if_available setting. */
if (config_lookup_string(config.cfg, "alsa.use_mmap_if_available", &str)) {
if (strcasecmp(str, "no") == 0)
config.no_mmap = 1;
else if (strcasecmp(str, "yes") == 0)
config.no_mmap = 0;
else
die("Invalid use_mmap_if_available option choice \"%s\". It should be \"yes\" or \"no\"");
}
/* Get the optional period size value */
if (config_lookup_int(config.cfg, "alsa.period_size", &value)) {
set_period_size_request = 1;
debug(1, "Value read for period size is %d.", value);
if (value < 0)
die("Invalid alsa period size setting \"%d\". It "
"must be greater than 0.",
value);
else
period_size_requested = value;
}
/* Get the optional buffer size value */
if (config_lookup_int(config.cfg, "alsa.buffer_size", &value)) {
set_buffer_size_request = 1;
debug(1, "Value read for buffer size is %d.", value);
if (value < 0)
die("Invalid alsa buffer size setting \"%d\". It "
"must be greater than 0.",
value);
else
buffer_size_requested = value;
}
}
optind = 1; // optind=0 is equivalent to optind=1 plus special behaviour
argv--; // so we shift the arguments to satisfy getopt()
argc++;
// some platforms apparently require optreset = 1; - which?
int opt;
while ((opt = getopt(argc, argv, "d:t:m:c:i:")) > 0) {
switch (opt) {
case 'd':
alsa_out_dev = optarg;
break;
case 't':
inform("The alsa backend -t option is deprecated and has been ignored. "
"FYI, using the -c option automatically chooses a hardware "
"mixer.");
break;
case 'm':
alsa_mix_dev = optarg;
break;
case 'c':
alsa_mix_ctrl = optarg;
hardware_mixer = 1;
break;
case 'i':
alsa_mix_index = strtol(optarg, NULL, 10);
break;
default:
help();
die("Invalid audio option -%c specified", opt);
}
}
if (optind < argc)
die("Invalid audio argument: %s", argv[optind]);
debug(1, "Output device name is \"%s\".", alsa_out_dev);
if (hardware_mixer) {
if (alsa_mix_dev == NULL)
alsa_mix_dev = alsa_out_dev;
// Open mixer
open_mixer();
if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, &alsa_mix_maxv) <
0)
debug(1, "Can't read mixer's [linear] min and max volumes.");
else {
if (snd_mixer_selem_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb, &alsa_mix_maxdb) ==
0) {
audio_alsa.volume = &volume; // insert the volume function now we know it can do dB stuff
audio_alsa.parameters = ¶meters; // likewise the parameters stuff
if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) {
// Raspberry Pi does this
debug(1, "Lowest dB value is a mute -- try minimum volume +1");
if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1,
&alsa_mix_mindb) != 0)
debug(1, "Can't get dB value corresponding to a minimum volume + 1.");
}
debug(1, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0,
(1.0 * alsa_mix_maxdb) / 100.0);
} else {
// use the linear scale and do the db conversion ourselves
debug(1, "note: the hardware mixer specified -- \"%s\" -- does not have "
"a dB volume scale, so it can't be used. Trying software "
"volume control.",
alsa_mix_ctrl);
if (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0)
die("Cannot open control \"%s\"", alsa_mix_dev);
if (snd_ctl_elem_id_malloc(&elem_id) < 0)
die("Cannot allocate memory for control \"%s\"", alsa_mix_dev);
snd_ctl_elem_id_set_interface(elem_id, SND_CTL_ELEM_IFACE_MIXER);
snd_ctl_elem_id_set_name(elem_id, alsa_mix_ctrl);
if (snd_ctl_get_dB_range(ctl, elem_id, &alsa_mix_mindb, &alsa_mix_maxdb) == 0) {
debug(1, "Volume control \"%s\" has dB volume from %f to %f.", alsa_mix_ctrl,
(1.0 * alsa_mix_mindb) / 100.0, (1.0 * alsa_mix_maxdb) / 100.0);
has_softvol = 1;
} else {
debug(1, "Cannot get the dB range from the volume control \"%s\"", alsa_mix_ctrl);
}
/*
debug(1, "Min and max volumes are %d and
%d.",alsa_mix_minv,alsa_mix_maxv);
alsa_mix_maxdb = 0;
if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0))
alsa_mix_mindb =
-20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0));
else if (alsa_mix_maxv!=0)
alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0);
audio_alsa.volume = &linear_volume; // insert the linear volume function
audio_alsa.parameters = ¶meters; // likewise the parameters stuff
debug(1,"Max and min dB calculated are %d and
%d.",alsa_mix_maxdb,alsa_mix_mindb);
*/
}
}
if ((config.alsa_use_playback_switch_for_mute==1) && (snd_mixer_selem_has_playback_switch(alsa_mix_elem))) {
audio_alsa.mute = &mute; // insert the mute function now we know it can do muting stuff
// debug(1, "Has mixer and mute ability we will use.");
} else {
// debug(1, "Has mixer but not using hardware mute.");
}
close_mixer();
} else {
// debug(1, "Has no mixer and thus no hardware mute.");
}
alsa_mix_handle = NULL;
pthread_mutex_unlock(&alsa_mutex);
return 0;
}
static void deinit(void) {
// debug(2,"audio_alsa deinit called.");
stop();
}
int open_alsa_device(void) {
const snd_pcm_uframes_t minimal_buffer_headroom =
352 * 2; // we accept this much headroom in the hardware buffer, but we'll
// accept less
const snd_pcm_uframes_t requested_buffer_headroom =
minimal_buffer_headroom + 2048; // we ask for this much headroom in the
// hardware buffer, but we'll accept less
int ret, dir = 0;
unsigned int my_sample_rate = desired_sample_rate;
// snd_pcm_uframes_t frames = 441 * 10;
snd_pcm_uframes_t buffer_size, actual_buffer_length;
snd_pcm_access_t access;
// ensure no calls are made to the alsa device enquiring about the buffer length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
// ensure no calls are made to the alsa device enquiring about the buffer length if
// synchronisation is disabled.
if (config.no_sync != 0)
audio_alsa.delay = NULL;
ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0)
return (ret);
// die("Alsa initialization failed: unable to open pcm device: %s.",
// snd_strerror(ret));
snd_pcm_hw_params_alloca(&alsa_params);
ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
if (ret < 0) {
die("audio_alsa: Broken configuration for device \"%s\": no configurations "
"available",
alsa_out_dev);
}
if ((config.no_mmap == 0) &&
(snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_MMAP_INTERLEAVED) >=
0)) {
if (output_method_signalled == 0) {
debug(1, "Output written using MMAP");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
alsa_pcm_write = snd_pcm_mmap_writei;
} else {
if (output_method_signalled == 0) {
debug(1, "Output written with RW");
output_method_signalled = 1;
}
access = SND_PCM_ACCESS_RW_INTERLEAVED;
alsa_pcm_write = snd_pcm_writei;
}
ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params, access);
if (ret < 0) {
die("audio_alsa: Access type not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
}
snd_pcm_format_t sf;
switch (sample_format) {
case SPS_FORMAT_S8:
sf = SND_PCM_FORMAT_S8;
break;
case SPS_FORMAT_U8:
sf = SND_PCM_FORMAT_U8;
break;
case SPS_FORMAT_S16:
sf = SND_PCM_FORMAT_S16;
break;
case SPS_FORMAT_S24:
sf = SND_PCM_FORMAT_S24;
break;
case SPS_FORMAT_S24_3LE:
sf = SND_PCM_FORMAT_S24_3LE;
break;
case SPS_FORMAT_S24_3BE:
sf = SND_PCM_FORMAT_S24_3BE;
break;
case SPS_FORMAT_S32:
sf = SND_PCM_FORMAT_S32;
break;
default:
die("Unsupported output format at audio_alsa.c");
}
ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sf);
if (ret < 0) {
die("audio_alsa: Sample format %d not available for device \"%s\": %s", sample_format, alsa_out_dev,
snd_strerror(ret));
}
ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
if (ret < 0) {
die("audio_alsa: Channels count (2) not available for device \"%s\": %s", alsa_out_dev,
snd_strerror(ret));
}
ret = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, &my_sample_rate, &dir);
if (ret < 0) {
die("audio_alsa: Rate %iHz not available for playback: %s", desired_sample_rate,
snd_strerror(ret));
}
if (set_period_size_request != 0) {
debug(1, "Attempting to set the period size");
ret = snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &period_size_requested,
&dir);
if (ret < 0) {
die("audio_alsa: cannot set period size of %lu: %s", period_size_requested,
snd_strerror(ret));
snd_pcm_uframes_t actual_period_size;
snd_pcm_hw_params_get_period_size(alsa_params, &actual_period_size, &dir);
if (actual_period_size != period_size_requested)
inform("Actual period size set to a different value than requested. Requested: %lu, actual "
"setting: %lu",
period_size_requested, actual_period_size);
}
}
if (set_buffer_size_request != 0) {
debug(1, "Attempting to set the buffer size to %lu", buffer_size_requested);
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params, &buffer_size_requested);
if (ret < 0) {
die("audio_alsa: cannot set buffer size of %lu: %s", buffer_size_requested,
snd_strerror(ret));
snd_pcm_uframes_t actual_buffer_size;
snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_size);
if (actual_buffer_size != buffer_size_requested)
inform("Actual period size set to a different value than requested. Requested: %lu, actual "
"setting: %lu",
buffer_size, actual_buffer_size);
}
}
ret = snd_pcm_hw_params(alsa_handle, alsa_params);
if (ret < 0) {
die("audio_alsa: Unable to set hw parameters for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
}
if (my_sample_rate != desired_sample_rate) {
die("Can't set the D/A converter to %d.", desired_sample_rate);
}
ret = snd_pcm_hw_params_get_buffer_size(alsa_params, &actual_buffer_length);
if (ret < 0) {
die("audio_alsa: Unable to get hw buffer length for device \"%s\": %s.", alsa_out_dev,
snd_strerror(ret));
}
if (actual_buffer_length < config.audio_backend_buffer_desired_length + minimal_buffer_headroom) {
/*
// the dac buffer is too small, so let's try to set it
buffer_size =
config.audio_backend_buffer_desired_length + requested_buffer_headroom;
ret = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params,
&buffer_size);
if (ret < 0)
die("audio_alsa: Unable to set hw buffer size to %lu for device \"%s\": "
"%s.",
config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
alsa_out_dev, snd_strerror(ret));
if (config.audio_backend_buffer_desired_length + minimal_buffer_headroom >
buffer_size) {
die("audio_alsa: Can't set hw buffer size to %lu or more for device "
"\"%s\". Requested size: %lu, granted size: %lu.",
config.audio_backend_buffer_desired_length + minimal_buffer_headroom,
alsa_out_dev, config.audio_backend_buffer_desired_length +
requested_buffer_headroom,
buffer_size);
}
*/
debug(1, "The alsa buffer is to small (%lu bytes) to accommodate the desired backend buffer "
"length (%ld) you have chosen.",
actual_buffer_length, config.audio_backend_buffer_desired_length);
}
if (alsa_characteristics_already_listed == 0) {
alsa_characteristics_already_listed = 1;
int log_level = 2; // the level at which debug information should be output
int rc;
snd_pcm_access_t access_type;
snd_pcm_format_t format_type;
snd_pcm_subformat_t subformat_type;
unsigned int val, val2;
unsigned int uval, uval2;
int sval;
int dir;
snd_pcm_uframes_t frames;
debug(log_level, "PCM handle name = '%s'", snd_pcm_name(alsa_handle));
// ret = snd_pcm_hw_params_any(alsa_handle, alsa_params);
// if (ret < 0) {
// die("audio_alsa: Cannpot get configuration for device \"%s\": no
// configurations
//"
// "available",
// alsa_out_dev);
// }
debug(log_level, "alsa device parameters:");
snd_pcm_hw_params_get_access(alsa_params, &access_type);
debug(log_level, " access type = %s", snd_pcm_access_name(access_type));
snd_pcm_hw_params_get_format(alsa_params, &format_type);
debug(log_level, " format = '%s' (%s)", snd_pcm_format_name(format_type),
snd_pcm_format_description(format_type));
snd_pcm_hw_params_get_subformat(alsa_params, &subformat_type);
debug(log_level, " subformat = '%s' (%s)", snd_pcm_subformat_name(subformat_type),
snd_pcm_subformat_description(subformat_type));
snd_pcm_hw_params_get_channels(alsa_params, &uval);
debug(log_level, " number of channels = %u", uval);
sval = snd_pcm_hw_params_get_sbits(alsa_params);
debug(log_level, " number of significant bits = %d", sval);
snd_pcm_hw_params_get_rate(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " rate = %u frames per second (<).", uval);
break;
case 0:
debug(log_level, " rate = %u frames per second (precisely).", uval);
break;
case 1:
debug(log_level, " rate = %u frames per second (>).", uval);
break;
}
if (snd_pcm_hw_params_get_rate_numden(alsa_params, &uval, &uval2) == 0)
debug(log_level, " precise (rational) rate = %.3f frames per second (i.e. %u/%u).", uval,
uval2, ((double)uval) / uval2);
else
debug(log_level, " precise (rational) rate information unavailable.");
snd_pcm_hw_params_get_period_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " period_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " period_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " period_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_period_size(alsa_params, &frames, &dir);
switch (dir) {
case -1:
debug(log_level, " period_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " period_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " period_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_buffer_time(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " buffer_time = %u us (<).", uval);
break;
case 0:
debug(log_level, " buffer_time = %u us (precisely).", uval);
break;
case 1:
debug(log_level, " buffer_time = %u us (>).", uval);
break;
}
snd_pcm_hw_params_get_buffer_size(alsa_params, &frames);
switch (dir) {
case -1:
debug(log_level, " buffer_size = %lu frames (<).", frames);
break;
case 0:
debug(log_level, " buffer_size = %lu frames (precisely).", frames);
break;
case 1:
debug(log_level, " buffer_size = %lu frames (>).", frames);
break;
}
snd_pcm_hw_params_get_periods(alsa_params, &uval, &dir);
switch (dir) {
case -1:
debug(log_level, " periods_per_buffer = %u (<).", uval);
break;
case 0:
debug(log_level, " periods_per_buffer = %u (precisely).", uval);
break;
case 1:
debug(log_level, " periods_per_buffer = %u (>).", uval);
break;
}
}
return (0);
}
static void start(int i_sample_rate, int i_sample_format) {
// debug(2,"audio_alsa start called.");
if (i_sample_rate == 0)
desired_sample_rate = 44100; // default
else
desired_sample_rate = i_sample_rate; // must be a variable
if (i_sample_format == 0)
sample_format = SPS_FORMAT_S16; // default
else
sample_format = i_sample_format;
}
int delay(long *the_delay) {
// snd_pcm_sframes_t is a signed long -- hence the return of a "long"
int reply;
// debug(3,"audio_alsa delay called.");
if (alsa_handle == NULL) {
return -ENODEV;
} else {
pthread_mutex_lock(&alsa_mutex);
int derr, ignore;
if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING) {
*the_delay = 0; // just to see what happens
reply = snd_pcm_delay(alsa_handle, the_delay);
if (reply != 0) {
debug(1, "Error %d in delay(): \"%s\". Delay reported is %d frames.", reply,
snd_strerror(reply), *the_delay);
ignore = snd_pcm_recover(alsa_handle, reply, 1);
}
} else if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) {
*the_delay = 0;
reply = 0; // no error
} else {
if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN) {
*the_delay = 0;
reply = 0; // no error
} else {
reply = -EIO;
debug(1, "Error -- ALSA delay(): bad state: %d.", snd_pcm_state(alsa_handle));
}
if ((derr = snd_pcm_prepare(alsa_handle))) {
ignore = snd_pcm_recover(alsa_handle, derr, 1);
debug(1, "Error preparing after delay error: \"%s\".", snd_strerror(derr));
}
}
pthread_mutex_unlock(&alsa_mutex);
// here, occasionally pretend there's a problem with pcm_get_delay()
// if ((random() % 100000) < 3) // keep it pretty rare
// reply = -EPERM; // pretend something bad has happened
return reply;
}
}
static void play(short buf[], int samples) {
// debug(3,"audio_alsa play called.");
int ret = 0;
if (alsa_handle == NULL) {
pthread_mutex_lock(&alsa_mutex);
ret = open_alsa_device();
if (ret == 0) {
if (audio_alsa.volume)
do_volume(set_volume);
if (audio_alsa.mute)
do_mute(0);
}
pthread_mutex_unlock(&alsa_mutex);
}
if (ret == 0) {
pthread_mutex_lock(&alsa_mutex);
snd_pcm_sframes_t current_delay = 0;
int err, ignore;
if (snd_pcm_state(alsa_handle) == SND_PCM_STATE_XRUN) {
if ((err = snd_pcm_prepare(alsa_handle))) {
ignore = snd_pcm_recover(alsa_handle, err, 1);
debug(1, "Error preparing after underrun: \"%s\".", snd_strerror(err));
}
}
if ((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
(snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)) {
if (buf == NULL)
debug(1, "NULL buffer passed to pcm_writei -- skipping it");
if (samples == 0)
debug(1, "empty buffer being passed to pcm_writei -- skipping it");
if ((samples != 0) && (buf != NULL)) {
err = alsa_pcm_write(alsa_handle, (char *)buf, samples);
if (err < 0) {
debug(1, "Error %d writing %d samples in play(): \"%s\".", err, samples,
snd_strerror(err));
ignore = snd_pcm_recover(alsa_handle, err, 1);
}
}
} else {
debug(1, "Error -- ALSA device in incorrect state (%d) for play.",
snd_pcm_state(alsa_handle));
if ((err = snd_pcm_prepare(alsa_handle))) {
ignore = snd_pcm_recover(alsa_handle, err, 1);
debug(1, "Error preparing after play error: \"%s\".", snd_strerror(err));
}
}
pthread_mutex_unlock(&alsa_mutex);
}
}
static void flush(void) {
// debug(2,"audio_alsa flush called.");
pthread_mutex_lock(&alsa_mutex);
int derr;
do_mute(1);
if (alsa_handle) {
// debug(1,"Dropping frames for flush...");
/*
if ((derr = snd_pcm_drop(alsa_handle)))
debug(1, "Error dropping frames: \"%s\".", snd_strerror(derr));
// debug(1,"Dropped frames ok. State is %d.",snd_pcm_state(alsa_handle));
if ((derr = snd_pcm_prepare(alsa_handle)))
debug(1, "Error preparing after flush: \"%s\".", snd_strerror(derr));
// debug(1,"Frames successfully dropped.");
*/
/*
if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED)
debug(1,"Flush returns to SND_PCM_STATE_PREPARED state.");
if (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING)
debug(1,"Flush returns to SND_PCM_STATE_RUNNING state.");
*/
/*
if (!((snd_pcm_state(alsa_handle) == SND_PCM_STATE_PREPARED) ||
(snd_pcm_state(alsa_handle) == SND_PCM_STATE_RUNNING)))
debug(1, "Flush returning unexpected state -- %d.", snd_pcm_state(alsa_handle));
*/
// this is derived from http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.html#a45
if ((derr=snd_pcm_nonblock(alsa_handle, 0)))
debug(1, "Error %d (\"%s\") unblocking output device.", derr, snd_strerror(derr));
if ((derr=snd_pcm_drain(alsa_handle)))
debug(1, "Error %d (\"%s\") draining output device.", derr, snd_strerror(derr));
if ((derr=snd_pcm_nonblock(alsa_handle, 1)))
debug(1, "Error %d (\"%s\") reblocking output device.", derr, snd_strerror(derr));
if ((derr=snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing output device hardware.", derr, snd_strerror(derr));
// flush also closes the device
snd_pcm_close(alsa_handle);
alsa_handle = NULL;
}
pthread_mutex_unlock(&alsa_mutex);
}
static void stop(void) {
// debug(2,"audio_alsa stop called.");
// when we want to stop, we want the alsa device
// to be closed immediately -- we may even be killing the thread, so we
// don't wish to wait
// so we should flush first
flush(); // flush will also close the device
// close_alsa_device();
}
static void parameters(audio_parameters *info) {
info->minimum_volume_dB = alsa_mix_mindb;
info->maximum_volume_dB = alsa_mix_maxdb;
}
void do_volume(double vol) { // caller is assumed to have the alsa_mutex when using this function
debug(3, "Setting volume db to %f.", vol);
set_volume = vol;
if (volume_set_request && open_mixer()) {
if (has_softvol) {
if (ctl && elem_id) {
snd_ctl_elem_value_t *value;
long raw;
if (snd_ctl_convert_from_dB(ctl, elem_id, (long)vol, &raw, 0) < 0)
debug(1, "Failed converting dB gain to raw volume value for the "
"software volume control.");
snd_ctl_elem_value_alloca(&value);
snd_ctl_elem_value_set_id(value, elem_id);
snd_ctl_elem_value_set_integer(value, 0, raw);
snd_ctl_elem_value_set_integer(value, 1, raw);
if (snd_ctl_elem_write(ctl, value) < 0)
debug(1, "Failed to set playback dB volume for the software volume "
"control.");
}
} else {
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 0) != 0) {
debug(1, "Can't set playback volume accurately to %f dB.", vol);
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, -1) != 0)
if (snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol, 1) != 0)
debug(1,"Could not set playback dB volume on the mixer.");
}
}
volume_set_request = 0; // any external request that has been made is now satisfied
// debug(1,"Alsa volume actually set.");
close_mixer();
}
}
void volume(double vol) {
pthread_mutex_lock(&alsa_mutex);
volume_set_request = 1; // an external request has been made to set the volume
do_volume(vol);
pthread_mutex_unlock(&alsa_mutex);
}
/*
static void linear_volume(double vol) {
debug(2, "Setting linear volume to %f.", vol);
set_volume = vol;
if (hardware_mixer && alsa_mix_handle) {
double linear_volume = pow(10, vol);
// debug(1,"Linear volume is %f.",linear_volume);
long int_vol = alsa_mix_minv + (alsa_mix_maxv - alsa_mix_minv) * linear_volume;
// debug(1,"Setting volume to %ld, for volume input of %f.",int_vol,vol);
if (alsa_mix_handle) {
if (snd_mixer_selem_set_playback_volume_all(alsa_mix_elem, int_vol) != 0)
die("Failed to set playback volume");
}
}
}
*/